Device and system for selective forwarding

ABSTRACT

A system and method of operating equipment and services to allow enhanced global transport of IP packets is presented. A global virtual network with guaranteed capacity is used to transport said IP packets. A number of application-specific forwarding devices are deployed to detect and forward selected traffic types to the virtual network. The application-specific selective forwarding devices can be implemented based on the IP packets analysis, or by deploying enhanced control protocols like SIP/H.323.

CROSS-REFERENCE TO RELATED APPLICATIONS

This is a continuation-in-part under 35 U.S.C. §120 of International Patent Application No. PCT/NO08/000469 filed Dec. 29, 2008, and a continuation-in-part under 35 U.S.C. §120 of International Patent Application No. PCT/NO08/000470 filed Dec. 29, 2008, and further claims priority under 35 U.S.C. §119 of each of Norwegian Patent Application No. 20080028 filed Jan. 2, 2008, and Norwegian Patent Application No. 20080027 filed Jan. 2, 2008. The disclosures of all of the foregoing applications are hereby incorporated herein by reference in their respective entireties, for all purposes.

BACKGROUND

In the use model that dominates the Internet today, communication equipment (1) that produces data streams is situated at the ends or edges of the network. The data are sent from the source endpoint to the destination endpoint using the network system. The network system comprises local networks on the source and destination side (13), one or more local Internet Service Providers (ISP, 10) and one or more transport networks (11) as shown in FIG. 1.

Local networks (13) implement secure communication environments typically with private addresses and a firewall toward the rest of the Internet. Local Internet service provider networks (10) aggregate a number of local networks and use the transport providers (11) to reach each other.

Data packets enter the network from the end systems typically using a socket interface. In this model, the socket is identified by the pentuple of (source address, destination address, source port number, destination port number, protocol type). The packets comprise two parts: a header and a payload. The header provides control information, while the payload contains higher-layer (e.g., application-level) data. The header identifies said pentuple.

The data are forwarded between the forwarding devices (20) hop-by-hop using the information in the packet header. Although additional functionality can be implemented in a network, only this hop-by-hop forwarding toward the destination is implemented universally in the Internet.

Therefore, the network functionality an endpoint and its associated local network components typically implement is related to network addressing, including

-   -   a) Resolving of symbolic names to network addresses; and     -   b) Support for traversal of Network Address Translator (NAT) and         firewall traversal.

This simple interface does not allow the endpoint to select the end-to-end network path. Only the destination can be selected, and the network itself chooses the path. Typically, the packet will have to traverse multiple administrative authorities on its path.

The Internet provides so-called best-effort service to its users. This means the packets are transported from node to node toward their destination. They can be temporarily stored in the transit nodes awaiting available network capacity to continue the journey (buffering). The nodes are free to discard any packet; this would typically happen if a node receives more packets than it can forward in the moment and its buffering capacity is exceeded. No notification is given to the sender of the packet.

Many applications are interactive or bandwidth-demanding and have special network requirements such as low latency or low packet loss. These applications include among others gaming, business information feeds and multimedia communications. The traditional Internet model is insufficient for these applications. Advanced endpoint equipment may experience reduced quality due to congestion and packet loss.

Remedies that can improve the network quality and availability include establishing private wide area networks, both physical and virtual. Large enterprises may rent or deploy network capacity to connect their locations within the enterprise, but with a significant cost. Furthermore, one can influence the packet path using overlay or peer-to-peer architectures. Overlays and peer-to-peer networks cannot improve the network quality unless the network resources are provided to them with capacity guarantees, which is typically not the case.

Recently, some Internet Service Providers (ISPs) have extended their network access and Virtual Private Networks (VPN) services by offering new network service models.

One model is to provide the infrastructure for temporary data storage (caching) to enhance large-scale one-to-many data streaming. This model conserves bandwidth, but it assumes delay-tolerance and is not suitable for interactive, real-time communications.

Another network service model is to provide infrastructure with guaranteed bandwidth to customers with geographically diverse office locations together with associated private network maintenance services. This service is different from VPNs in that there can be given bandwidth and latency guarantees to the traffic, and the customer need not have maintenance personnel employed. The model is however limited to the locations where the provider has physical infrastructure available, and cannot be extended to arbitrary communication peers.

Some providers build their business model on extensive Border Gateway Protocol (BGP) peering with locally present ISPs, enhancing the performance of their hosting services. This method improves the network service quality only locally.

Finally, some providers rent network capacity or deploy it themselves where needed and provide end-to-end guaranteed bandwidth service. This approach provides excellent network infrastructure, but is coarse and expensive and therefore reserved only to the most well-funded enterprise customers. Communication with arbitrary peers is not possible.

Quality of Service in the Internet

Recognizing the shortcomings of the best effort Internet service, the research community has proposed Quality of Service (QoS) models for the Internet.

In IntServ [RFC1633], network resources can be reserved end-to-end using a special signaling protocol called RSVP. The resources are reserved per flow, along the standard routing path. Such per-flow reservations scale poorly in the Internet where millions of flows are running concurrently, and are seldom deployed in practice.

DiffServ [RFC2475] is a QoS framework for differentiation between different traffic classes. DiffServ scales well and can be used to provide a better service to a certain segments of the network traffic like VoIP. However, DiffServ provides no hard QoS guarantees, only prioritizing a given traffic type in front of another.

Both Internet QoS service frameworks have a serious shortcoming in that they need service agreements between the administrative domains. Furthermore, the local ISP would have to guarantee the service level to the user, without having full control of how the data is sent to the destination. This has been proved to be difficult in practice, and IntServ and DiffServ remain used largely in private networks.

Network Virtualization

Network virtualization has recently been proposed as a means of deploying global network services. In a recent architecture proposal called CABO (“Concurrent Architectures are Better than One” [CABO]), virtual network links connect virtual routers to deploy a range of concurrent internets. The virtual links are implemented using any of many available technologies including MPLS and IP tunneling. Virtual routers are running as processes on real routing equipment owned and managed by the infrastructure providers. The virtual routers have their integral resources such as output queues and schedulers. There is also a possibility to connect virtual links based on infrastructure owned by different providers using the virtual routers.

The distinction between the infrastructure providers and the network service providers is a novel concept in CABO and facilitates implementation of custom global network services. These could include secure networks, QoS networks, and networks with different addressing and routing schemes providing yet unknown services. CABO also advocates deployment of a signaling system for dynamic establishment of virtual network topologies. However, current business and security models in the Internet do not encourage deployment of CABO since the network operators do not accept third-party access to their critical infrastructure.

Real-Time Conferencing

Real-time multimedia conferencing has gained substantial popularity recently, particularly in audio (telephony/Voice over IP) applications. Typically, a control system based on the Session Initiation Protocol, H.323 or a proprietary protocol (e.g. Skype) establishes a connection between the endpoints. The endpoints encode the media (audio, video, text for short messages, etc.) and send them as IP packets.

FIG. 2 shows a typical state of the art deployment for VoIP and/or video conferencing using SIP or H.323. The VoIP/conferencing operator typically operates a control infrastructure (17) with at least a control server (30) and optionally media gateways (22) to facilitate audio communication from IP-based networks (11,13,14) to PSTN (18). The communication is initiated in a terminal connected to a terminal adapter (32). Many terminals include the terminal adapter capabilities inside the terminal, constituting a multimedia terminal, often called a “softphone” if implemented inside a mobile phone/computer. The call is signaled from the terminal to the control server (30) using a control protocol (220). This control server is typically a SIP or H.323 proxy, which communicate with the remote terminal and establish a connection. The terminals are instructed to use an IP/port address combination for the media communication. The media stream (250) takes the default path through the IP network (11,13,14) to the destination terminal, and voice or video communication can begin. If the called party is a terminal on the PSTN network, the control server (30) will direct the call to a suitable media gateway (22) and complete the call over regular PSTN.

NAT and Firewall Traversal

Network Address Translation (NAT) devices perform translation of IP addresses between networks. For example, internal IP addresses on a user network (13) can have a mapping in the public Internet address space. In each packet leaving the local network (13) the IP header has to be modified by swapping the local addresses with the global ones. There are 4 basic types of NAT [STUN]:

-   -   Full Cone (not frequently used due to security issues)     -   Restricted Cone (the NAT IP mapping is only valid with outgoing         traffic to the destination)     -   Port Restricted Cone (same as Restricted Cone, but with IP and         port mapping)     -   Symmetric (different mapping for different destination         addresses)

Firewalls are used to enforce security in the local network. They typically close a majority of network ports and discard packets addressed to these ports.

NAT and firewalls are widely used. Any device operating in the Internet today must be capable of handling them. There are many practical solutions to the NAT and firewall problem, among these:

-   -   Universal Plug and Play (security issues by client control of         firewall pinholes, and does not work with cascading NATs).     -   STUN [STUN], which is a client-server system where the server         answers the clients query by embedding the perceived global         address of the client in the payload of its reply.     -   TURN server [TURN], which is a STUN server with additional         functionality to forward data packets to a given global IP         destination.     -   Application Level Gateway (ITU standard H.460.17/18/19) for use         with the H.323 standards.

Note that the present invention does not specify any particular of the NAT and firewall traversal solutions, and is intended to work with all of the above.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a schematic depicting a network system comprising local networks on the source and destination sides, one or more local Internet Service Providers, and one or more transport networks.

FIG. 2 is a schematic of a deployment for VOIP and/or video conferencing using SIP or H.323.

FIG. 3 is a schematic showing an application-specific forwarding device controlled by a control server according to one embodiment of the present invention.

FIG. 4 is a schematic showing an application-specific forwarding device based on the endpoint control system according to one embodiment of the present invention.

FIG. 5 is a schematic showing an application-specific forwarding device with data-driven operation according to one embodiment of the present invention.

FIG. 6 is a schematic showing dedicated transport network architecture according to one embodiment of the present invention.

FIG. 7 is a schematic showing a system according to one embodiment of the present invention.

FIG. 8 is a schematic showing a system with multiple endpoints per local network according to one embodiment of the present invention.

FIG. 9 is a schematic showing a system according to another embodiment of the present invention.

FIG. 10 is a schematic showing a system according to another embodiment of the present invention.

DETAILED DESCRIPTION OF THE INVENTION

The present invention enables the automatic redirection of traffic to a dedicated transport network by means of an application-specific selective data packet forwarding device. The device can be integrated in the communication terminal, placed in the local network (CPE), in equipment in an enterprise DMZ or using a public server. The device handles data packets, typically in RTP format, and must be present in the media data path between the communicating peers.

Features of embodiments of the invention include that:

-   -   It provides a targeted service for a specific data type (e.g.,         video, audio, game short messages, stock info)     -   It redirects the selected traffic to a dedicated transport         network.     -   It integrates AAA (Accounting, Authentication, Authorization) as         part of the system, in order to provide commercial services.     -   The application-specific selective packet-forwarding device         integrates with standard IP networks and standard communication         protocols like SIP and H.323.     -   Data packet redirection can be either control driven or data         driven:         -   Control-driven where a control protocol such as SIP, H.323             or another control protocol controls the             application-specific forwarding device and instructs it             which packets to redirect to the dedicated transport network             and which packets to forward over the shared Internet.         -   Data-driven where the application-specific forwarding device             analyzes data packets and decides which packets to redirect             to the dedicated transport network and which packets to             forward over the shared Internet.     -   Global IP addresses are mapped to the dedicated network PoPs and         IP addresses. This way the dedicated network can hold the         packets for as long as it takes to forward them to the PoP         nearest to their destination.

Application-Specific Packet-Forwarding Device

In one set of embodiments, application-specific packet forwarding devices (31) distinguish between the traffic that should be forwarded over the dedicated transport network (15) and the traffic that should be forwarded over the regular, shared Internet (11). The device comprises a packet filter (60) that classifies the data packets into at least two classes; one for the regular, shared Internet (11) and one for the dedicated transport network (15). The packet header (130) can be changed to reflect the intended transport. The packet payload (150) can be transported unaltered.

Latency or bandwidth sensitive traffic is thus redirected to a dedicated network (15) in order to provide certain QoS guarantees to the user of the network service. Said dedicated network is a transport network comprising retransmission devices (20) at multiple points of presence (PoPs), internally connected by guaranteed bandwidth channels (50), owned or rented from transport providers (12). The network is connected to multiple ISPs (10).

The traffic is redirected to the said dedicated transport network (15) using an application-specific selective forwarding device (31). Said device forwards selected traffic to any destination address to the address of a retransmission device (20) in the transport network. The traffic can be selected by analyzing the packet header (130) or packet content (150), or by interaction with a centralized or distributed server (30). To forward packets to the re-transmission device, said application-specific selective forwarding device can use many mechanisms, including IP tunnels and transport-level proxy mechanisms.

The packets selected for forwarding over the dedicated transport network (15) are addressed (102) to a retransmission device (20) within the dedicated network (15). The address of said re-transmission device (20) is determined by a mapping between an address space and the addresses of the re-transmission devices within the transport network. Said address space can be the IP address space, or the PSTN E.164 address space, or another address space that contains the network addresses of the call source and destination. Said re-transmission device (20) can be an IP router, or a transport relay (33) operating as a SIP or H.323 media proxy.

Said mapping can be implemented in several ways. It can be based on BGP routing information collected in the transport network from the connected ISPs. It can be based on measurements such as the current network load in the transport network. It can be based on the network distance, i.e., which re-transmission device is closest to the destination address, or which re-transmission device is closest to the source address.

There are two possible operation modes for the application-specific packet-forwarding device:

-   1) control protocol integration -   2) data-driven operation.

Device Embodiment 1 Control Protocol Integration

In this embodiment, the control system establishes the session using e.g. SIP as shown in FIG. 3. It then uses a control interface to instruct the application-specific forwarding device (31) which packets should be forwarded to the dedicated transport network (15) and which need not.

The communication between the control server (30) and the application-specific forwarding device (31) can be organized as a query-response protocol (200). The device (31) can ask the server (30) whether the packet with header field combination (101, 120) should be transported using the dedicated network or not. The server (30) can answer with the re-transmission device (20) or transport relay (33) address (102).

Packets sent over the dedicated transport network (15) are encapsulated to a given address (102) within the dedicated network (15). There, they are de-capsulated and forwarded to their destination (101).

The selective forwarding device (31) learns the mapping between the global addresses (101) and the local addresses (102) from the control servers (30) using a query-reply protocol (200). The device (31) can ask which address in the dedicated transport network (15) should be used to send data to destination (101) using message (210). The answer is provided in the message (211).

Alternatively the endpoint control system (32) can control the application-specific forwarding device based on the user configuration data or a management system data using a specialized protocol (210) as shown in FIG. 4.

Device Embodiment 2 Data Driven Operation

In this embodiment, shown in FIG. 5, the application-specific forwarding device is in the data path and it monitors all IP data packets it forwards using a packet analyzer (61). In the packet filter (60) it can select the packets that should be transported over the dedicated network (15) using header analysis or it quantifies the streams by associating the packets to the quintuple of (source address, destination address, source port, application port, protocol), possibly using wildcards on one or more fields, and forwards long-lasting or voluminous or otherwise selected streams over the dedicated network (15).

System Description

The system relies on a global dedicated transport network (15) as shown in FIG. 6, with the following properties:

-   -   The network has multiple Points of Presence (PoP), located in         vicinity of the endpoints operated by the service users.     -   The PoPs are connected using virtual or real network lines with         guaranteed bandwidth (50) leased from the transport providers         (12).     -   Access to the network is strictly controlled and typically         allowed to the paying customers only.

System Embodiment 1

The first embodiment comprises a dedicated transport network (15), an application-specific forwarding device (31) and compliant re-transmission devices (20), as shown in FIG. 7.

The dedicated transport network (15) comprises points of presence (PoP) that are located close to the local ISPs (10). The PoPs are interconnected using physical or virtual links (50) with guaranteed bandwidth.

This embodiment relies on an application-specific forwarding device (31) that selects relevant traffic from the local network (13) and forwards this traffic onto a dedicated transport network (15) via the standard ISP used by the customer (10). The forwarding can be done using IP tunneling or a proxy operation using SIP or H.323. The retransmission devices (20) can be implemented as IP routers or as SIP/H.323 media proxies.

Using this configuration, all unrelated traffic such as email and file transfer will go over the default IP route (11), while latency and bandwidth-sensitive traffic will be redirected to a dedicated transport network (15) with sufficient capacity as shown in FIG. 6.

The redirection can also be applied for signaling traffic, but the signaling can also use the regular IP route since it is not latency and bandwidth sensitive.

Optional control servers (30) may communicate with the application-specific forwarding device to assist in determination of the route selection. Additionally, they can be used for admission control, AAA and directory services.

In addition, each local network can have multiple endpoints, as shown in FIG. 8. Multiple endpoints can be served by a concentrator typically located in the enterprise DMZ.

The architecture supports multicast. Multi-party conferences can be arranged by unicast-multicast reflectors [REFLECT] deployed in (20) and native multicast deployed in (15).

System Embodiment 2

In this embodiment shown in FIG. 9, the control server (30) is located in the public IP network, typically close to the dedicated transport network. One or more transport relays (33) are placed centrally in the dedicated transport network and can serve many users.

In this embodiment, the SIP/H.323 endpoint control system (32) in the terminals (1) must be configured to always contact the control server (30) as outbound signaling proxy. Signaling path is indicated between the terminals and the proxy (220). The control server maintains a mapping in the transport relay (33) using a control protocol (200).

After initial signaling between the endpoint control system (32) and the control server (30), the terminal is instructed to send the media stream (250) over the dedicated transport network (15) using the transport relay (33) as outbound media proxy.

The control server rewrites source and destination IP addresses/ports as part of the media redirection, to ensure that a packet redirected to the media proxy will be forwarded on to the original destination.

System Embodiment 3

Embodiment 3 as shown in FIG. 10 is similar to embodiment 2, the difference being that the control server (30) does not control the transport relay (33) directly. Instead, a modified Interactive Connectivity Establishment [ICE] procedure is used to instruct the transport relay (33) where to send the packets.

In this embodiment, the functionality of the transport relay (33) is similar to that of a TURN server [TURN]. The caller endpoint control system (32) is configured to use the transport relay (33) as the outbound media proxy. Thus, it always directs media to the dedicated network. The modified Interactive Connectivity Establishment [ICE] procedure is used to avoid sending data between collocated endpoint via the dedicated transport network (15). This procedure includes two steps:

-   -   a) The caller endpoint verifies whether the called endpoint has         an IP address in the same network segment and that the called         endpoint can be contacted. If confirmative, no dedicated network         (15) is used.     -   b) Otherwise, the transport relay (33) is used.

When the transport relay (33) is used to transfer the data over the dedicated network (15), the NAT traversal is implicitly solved using the TURN technology.

REFERENCES

-   [CABO] Nick Feamster, Lixin Gao and Jennifer Rexford: “How to lease     Internet in your spare time”, ACM SIGCOMM Computer Communications     Review journal, pages 61-64, January 2007. -   [ICE] Rosenberg, J., “Interactive Connectivity Establishment (ICE):     A Protocol for Network Address Translator (NAT) Traversal for     Offer/Answer Protocols”, IETF draft-ietf-mmusic-ice-16 (work in     progress), June 2007. -   [REFLECT] Tarik Cicic, Haakon Bryhni, Steinar Sørlie:     “Multicast-Unicast Reflector”, In proceedings of Protocols for     Multimedia Communications (PROMS) conference, pages 60-69, Krakow,     Poland, 2000. -   [RFC1633] R. Braden, D. Clark, S. Shenker: “Integrated Services in     the Internet Architecture: an Overview”, IETF, June 1994. -   [RFC2475] S. Blake, D. Black, M. Carlson, E. Davies, Z. Wang, W.     Weiss “An Architecture for Differentiated Services”, IETF, December     1998. -   [RFC4364] E. Rosen and Y. Rekhter: “BGP/MPLS IP Virtual Private     Networks (VPNs)”, IETF, February 2006. -   [STUN] Rosenberg, J., Weinberger, J., Huitema, C. and R. Mahy,     “STUN—Simple Traversal of User Datagram Protocol (UDP) Through     Network Address Translators (NATs)”, RFC 3489, March 2003. -   [TURN] J. Rosenborg, R. Mahy, C. Huitema: “Traversal Using Relays     around NAT (TURN): Relay Extensions to Session Traversal Utilities     for NAT (STUN)”, IETF draft-ietf-behave-turn-06 (work in progress),     2007. 

What is claimed is:
 1. An audio- or video-conferencing transport system, comprising: a transport network, comprising multiple points of presence, connected by guaranteed bandwidth channels; a media transport relay in the transport network; a control server configured to provide admission control to the transport network; a first Internet Service Provider (ISP) network, connected to the transport network; a second ISP network, connected to the transport network; a first audio- or video-conferencing terminal, connected to the first ISP network; and a second audio- or video-conferencing terminal, connected to the second ISP network; wherein the first audio- or video-conferencing terminal is configured to send real-time audio or video data to the second audio- or video-conferencing terminal by addressing the real-time audio or video data to the media transport relay in the transport network, so as to cause the real-time audio or video data (i) to travel from the first audio- or video-conferencing terminal to the transport network via the first ISP network, and then (ii) when admitted by the control server, to travel through the transport network over a guaranteed bandwidth channel of the guaranteed bandwidth channels, and then (iii) to travel from the transport network to the second audio- or video-conferencing terminal via the second ISP network.
 2. The audio- or video-conferencing transport system of claim 1, wherein the first audio- or video-conferencing terminal is connected to the first ISP network via a local network, such that the real-time audio or video data will travel from the first audio- or video-conferencing terminal to the first ISP network via the local network.
 3. The audio- or video-conferencing transport system of claim 1, wherein the transport network comprises a plurality of media transport relays having respective IP addresses, and wherein the first audio- or video-conferencing terminal is configured to address the real-time audio or video data to an IP address of one media transport relay of the plurality of media transport relays in the transport network.
 4. The audio- or video-conferencing transport system of claim 1, wherein the transport network comprises a plurality of media transport relays, and wherein the first audio- or video-conferencing terminal is configured to address the real-time audio or video data to a media transport relay of the plurality of media transport relays in the transport network that is closest to the second audio- or video-conferencing terminal.
 5. The audio- or video-conferencing transport system of claim 1, wherein the transport network comprises a plurality of media transport relays, and wherein the first audio- or video-conferencing terminal is configured to address the real-time audio or video data to a media transport relay of the plurality of media transport relays in the transport network that is closest to the first audio- or video-conferencing terminal.
 6. The audio- or video-conferencing transport system of claim 1, wherein the first audio- or video-conferencing terminal is configured to use tunnelling to send the real-time audio or video data to the media transport relay in the transport network.
 7. The audio- or video-conferencing transport system of claim 1, wherein the first audio- or video-conferencing terminal is configured to use a proxy mechanism to send the real-time audio or video data to the media transport relay in the transport network.
 8. The audio- or video-conferencing transport system of claim 1 wherein the first audio- or video-conferencing terminal is configured to use TURN to send the real-time audio or video data to the media transport relay in the transport network.
 9. The audio- or video-conferencing transport system of claim 1, wherein the transport network comprises a plurality of media transport relays and wherein at least one media transport relay of the plurality of media transport relays is a SIP or H.323 media proxy.
 10. The audio- or video-conferencing transport system of claim 1, wherein the second audio- or video-conferencing terminal is connected to the second ISP network via a local network, such that the real-time audio or video data will travel from the second ISP network to the second audio- or video-conferencing terminal via the local network.
 11. The audio- or video-conferencing transport system of claim 1, wherein the transport network is configured to hold the real-time audio or video data for as long as necessary to forward the real-time audio or video data to a point of presence of the multiple points of presence in the transport network that is nearest to the second audio- or video-conferencing terminal.
 12. The audio- or video-conferencing transport system of claim 1, wherein the transport network comprises a plurality of media transport relays, wherein the plurality of media transport relays comprise unicast-multicast reflectors, and wherein the transport network supports multicast for enabling multi-party conferences.
 13. A method of transporting audio- or video-conferencing data utilizing an audio- or video-conferencing transport system of claim 1, the method comprising a first audio- or video-conferencing terminal connected to a first ISP network: sending authorization information to a control server in order to gain admission to a transport network comprising multiple points of presence, connected by guaranteed bandwidth channels; and sending real-time audio or video data to a second audio- or video-conferencing terminal connected to a second ISP network, and addressing the real-time audio or video data to a media transport relay in the transport network, to cause the real-time audio or video data (i) to travel from the first audio- or video-conferencing terminal to the transport network via the first ISP network, and then (ii) to travel through the transport network over a guaranteed bandwidth channel of the guaranteed bandwidth channels, and then (iii) to travel from the transport network to the second audio- or video-conferencing terminal via the second ISP network.
 14. The method of claim 13, wherein the media transport relay to which the real-time audio or video data is addressed is a SIP or H.323 media proxy.
 15. The method of claim 13, wherein the first audio- or video-conferencing terminal is connected to the first ISP network via a local network, and wherein the real-time audio or video data travels from the first audio- or video-conferencing terminal to the first ISP network via the local network.
 16. The method of claim 13, wherein the second audio- or video-conferencing terminal is connected to the second ISP network via a local network, and wherein the real-time audio or video data travels from the second ISP network to the second audio- or video-conferencing terminal via the local network.
 17. The method of claim 13, comprising the transport network forwarding the real-time audio or video data to a point of presence of the multiple points of presence in the transport network that is nearest to the second audio- or video-conferencing terminal, and holding the real-time audio or video data in the transport network for as long as necessary to carry out said forwarding.
 18. The method of claim 13, wherein the transport network comprises a plurality of media transport relays, and wherein the first audio- or video-conferencing terminal addresses the real-time audio or video data to a media transport relay of the plurality of media transport relays in the transport network that is closest to the first audio- or video-conferencing terminal.
 19. The method of claim 13, wherein the transport network comprises a plurality of media transport relays, and wherein the first audio- or video-conferencing terminal addresses the real-time audio or video data to a media transport relay of the plurality of media transport relays in the transport network that is closest to the second audio- or video-conferencing terminal.
 20. The method of claim 13, wherein the transport network comprises a plurality of media transport relays, and wherein the plurality of media transport relays comprise unicast-multicast reflectors, the method further comprising using multicast in the transport network to send the real-time audio or video data to multiple recipients.
 21. The method of claim 13, comprising the first audio- or video-conferencing terminal addressing the real-time audio or video data to an IP address of the media transport relay.
 22. The method of claim 13, comprising the first audio- or video-conferencing terminal using tunnelling to send the real-time audio or video data to the media transport relay in the transport network.
 23. The method of claim 13, comprising the first audio- or video-conferencing terminal using a proxy mechanism to send the real-time audio or video data to the media transport relay in the transport network.
 24. The method of claim 13, comprising the first audio- or video-conferencing terminal using TURN to send the real-time audio or video data to the media transport relay in the transport network. 